Avaya 1200-Series Application Note

Brand: Avaya

Model: 1200-Series

Category: Networking

Type: Application Note

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

1 of 44

11xx12xx-SM

Avaya Solution & Interoperability Test Lab

Application Notes for Avaya 1100- and 1200-Series IP

Deskphones R3.2 with Avaya Aura™ Communication

Manager R6, Avaya Aura™ Session Manager R6, and

Avaya Modular Messaging R5.2 – Issue 1.0

Abstract

These Application Notes describe a solution comprised of Avaya Aura™ Communication

Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and

1200-Series IP Deskphones with SIP software. During testing, the IP Deskphones successfully

registered with Session Manager, placed and received calls to and from SIP and non-SIP

telephones, and executed other telephony features such as conference, transfer, hold, and Off-

PBX-Station (OPS) related features such as Call Pickup, Call Park, Whisper Page, and

Transfer to Voice Mail.

Information in these Application Notes has been obtained through interoperability testing and

additional technical discussions, and was conducted at the Avaya Solution and Interoperability

Test Lab at the request of Avaya 1100- and 1200-Series IP Deskphone Product Management.

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

2 of 44

11xx12xx-SM

1. Introduction

These Application Notes describe a solution comprised of Avaya Aura™ Communication

Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and

1200-Series IP Deskphones with SIP software (formerly known as Nortel 1100- and 1200-Series

SIP Phones). These telephones were originally developed under the Nortel brand, and as such do

not currently support the Avaya Advanced SIP Telephony (AST) protocol implemented in Avaya

9600 Series IP Telephones (SIP). Nevertheless, Communication Manager and Session Manager

have the capability to extend some advanced telephony features to non-AST telephones. The

configuration steps described include how to set up these features as well as the standard calling

features supported by the phones. See Section 4 for a summary of the features supported.

2. Reference Configuration

In the test configuration shown below, the Avaya S8800 Server with Avaya G450 Media

Gateway is configured as an Evolution Server (CM-ES), and supports all of the telephones

shown. The SIP telephone models tested included the 1120E (4 line monochrome), 1140E (6

line monochrome), 1165E (8 line color), 1220 (4 line monochrome), and the 1230 (10 line

monochrome) running SIP firmware. The phones are directly registered to Session Manager and

are supported by Communication Manager configured as an Evolution Server (CM-ES).

Communication between Communication Manager and Avaya Modular Messaging is via

Session Manager, which uses an adaptation module to translate subscriber numbers between the

5-digit extensions used by Communication Manager and the normalized 11-digit numbers used

by Modular Messaging. Modular Messaging supports all telephones for voice messaging.

Figure 1: Sample Configuration

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

3 of 44

11xx12xx-SM

In general, a SIP telephone originates a call by sending a call request (SIP INVITE message) to

Session Manager, which then routes the call over a SIP trunk to Communication Manager for

origination services. If the call is destined for another local SIP telephone, Communication

Manager routes the call back over the SIP trunk to Session Manager for delivery to the

destination SIP telephone. If the call is destined for an H.323 or Digital telephone, then

Communication Manager terminates the call directly.

These application notes assume that Communication Manager and Session Manager are already

installed and basic configuration steps have been performed. Only steps relevant to SIP

telephone calling features will be described in this document. For further details on configuration

steps not covered in this document, consult the appropriate document in Section 10.

3. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration

provided.

Equipment Software/Firmware

Avaya S8800 Server with G450 Media

Gateway

Avaya Aura™ Communication Manager 6.0

Service Pack 0 (Load 345, Update 18246)

Avaya S8800 Server

Avaya Aura™ Session Manager 6.0, Load

600020

Avaya Aura™ System Manager 6.0, Load

600020

Avaya 9630 IP Telephone (SIP) 2.6.0.0

Avaya 9630 IP Telephone (H.323) 3.1.1

Avaya 1603 IP Telephone (SIP) R1.0.0

Avaya 6408D+ Digital Telephone -

Modular Messaging Storage Server 5.2, Service Pack 3 Patch 1

Modular Messaging Application Server 5.2, Service Pack 3 Patch 1

Avaya 1100-series IP deskphones (SIP) 03.02.15.05

Avaya 1200-series IP deskphones (SIP) 03.02.15.05

Table 1: Equipment and Software/Firmware

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

4 of 44

11xx12xx-SM

4. Calling Features

4.1. Overview

Table 2 below shows the calling features successfully tested. Notes on specific feature

operations are included in Section 4.2. In addition to basic calling capabilities, the Internet

Engineering Task Force (IETF) has defined a supplementary set of calling features in RFC 5359

[13], previously referred to as the SIPPING features. This provides a useful framework to

describe product capabilities and compare features supported by various equipment vendors.

Communication Manager can support many of these features if the telephone can not locally

support them. In addition, features beyond those specified in RFC 5359 can be extended to the

telephone using Communication Manager configured as an Evolution Server.

SUPPORTED FEATURES COMMENTS

Basic Calling features

Extension to extension call

Basic call to non-SIP phones

Intercept tones/displays Reorder with message

Mute

Redial

Call Waiting

Do Not Disturb Section 4.2.1

Speed Dial buttons

Redial from call logs

Compressed codecs G.729A, G.729AB, G.722-64k

Message Waiting Support

SIPPING (RFC 5359) Features

Call Hold

Consultation Hold

Music on Hold

Unattended Transfer

Attended Transfer

Call Forward Unconditional Sections 4.2.2, 5.8

Call Forward Busy Via FNE (Section 5.8)

Call Forward No Answer Via FNE (Section 5.8)

3-way conference - 3rd party added

3-way conference - 3rd party joins

Find-Me Modular Messaging “Find Me” feature

Incoming Call Screening Via Class Of Restriction (Section 5.9)

Outgoing Call Screening Via Class Of Restriction (Section 5.9)

Call Park/Unpark Via FNE (Section 5.8)

Call Pickup Via FNE (Section 5.8)

Additional Station-Side Features

Calling Name/Number Block Via FNE (Section 4.2.3, 5.8, 7.4)

Directed Call Pick-Up Via FNE (Section 5.8)

Priority Call Via FNE (Sections 4.2.4, 5.8, 5.9)

Transfer to Voice Mail Via FNE (Section 5.8)

Whisper Page Via FNE (Section 5.8)

Table 2: SIP Telephony Feature Support

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

5 of 44

11xx12xx-SM

Some supported features shown in Table 2 can be invoked by dialing a Feature Name Extension

(FNE). Or, a speed dial button on the telephone can be programmed to an FNE. Communication

Manager automatically handles many other standard features such as call coverage, trunk

selection using Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS),

Class Of Service/Class Of Restriction (COS/COR), and voice messaging. Details on operation

and administration for Communication Manager can be found in References [4-6].

4.2. Operational Notes

4.2.1. Do Not Disturb

When Do Not Disturb is activated, the call is not presented to or displayed on the phone. The

call follows the coverage path configured for the extension in Communication Manager. This

feature is locally supported by the telephone, and is recommended instead of the Communication

Manager FNEs for Send All Calls and Send All Calls Cancel.

4.2.2. Call Forward Unconditional

It is recommended that this feature be administered as a Communication Manager FNE rather

than using the local call forward of the telephone. The user of local call forward will not benefit

from any of the call coverage features available in Communication Manager, including coverage

to voice messaging.

4.2.3. Calling Name/Number Block

The Avaya 1100- and 1200-Series IP Deskphones support privacy by means of the Remote-

Party-ID SIP header in the INVITE message. Since Communication Manager supports the

newer Privacy header along with P-Asserted-Identity header, this local feature is not supported.

It is recommended that the Calling Number Block FNE in Communication Manager be used

instead. This can be configured as speed dial button on the telephone (see Sections 5.8 and 7.4).

4.2.4. Priority Call

The telephone may originate priority calls based on the class of service administered for it (see

Section 5.9) or if the user dials the appropriate FNE. Note however, that it will not indicate a

received priority call. Avaya 9600 Series IP Telephones (SIP, H.323) and Avaya 6408D+

Digital Telephones will properly indicate them via distinctive ringing and calling party display.

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

6 of 44

11xx12xx-SM

5. Configure Avaya Aura™ Communication Manager

This section describes a procedure for setting up a SIP trunk between Communication Manager

serving as an Evolution Server and Session Manager. This includes steps for setting up IP

codecs, an IP network region, SIP signaling group, SIP trunk group, dial plan, class of service,

class of restriction, and call routing. Also, a procedure is described here to configure SIP

telephones and features available with OPS in Communication Manager. Configuration in the

following sections focuses on the fields where a value needs to be entered or modified. Default

values are used for all other fields.

These steps are performed from the Communication Manager System Access Terminal (SAT)

interface. Avaya and third party SIP telephones are configured as Off-PBX Stations (OPS) in

Communication Manager. Communication Manager does not directly control an OPS endpoint,

but its features and calling privileges can be applied to it by associating a local extension with

the OPS endpoint. Similarly, a SIP telephone in Session Manager is associated with an extension

on Communication Manager. SIP telephones register with Session Manager and use

Communication Manager for call origination and termination services, including Feature Name

Extension (FNE) support. Enter the save translation command after completing this section.

5.1. Capacity Verification

Before a SIP trunk or OPS endpoints can be configured, it is necessary to verify if there is

enough capacity.

Step Description

Enter the display system-parameters customer-options command. Verify that there are

sufficient Maximum Off-PBX Telephones – OPS licenses. If not, contact an authorized

Avaya account representative to obtain additional licenses.

display system-parameters customer-options Page 1 of 11

OPTIONAL FEATURES

G3 Version: V16 Software Package: Enterprise

Location: A System ID (SID): 1

Platform: 28 Module ID (MID): 1

USED

Platform Maximum Ports: 65000 296

Maximum Stations: 36000 124

Maximum XMOBILE Stations: 41000 0

Maximum Off-PBX Telephones - EC500: 36000 1

Maximum Off-PBX Telephones - OPS: 36000 101

Maximum Off-PBX Telephones - PBFMC: 36000 0

Maximum Off-PBX Telephones - PVFMC: 36000 0

Maximum Off-PBX Telephones - SCCAN: 0 0

Maximum Survivable Processors: 313 0

(NOTE: You must logoff & login to effect the permission changes.)display system-

parameters customer-options

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

7 of 44

11xx12xx-SM

Proceed to Page 2 of the OPTIONAL FEATURES form. Verify that the number of

Maximum Administered SIP Trunks supported by the system is sufficient for the

number of SIP trunks needed. If not, contact an authorized Avaya account representative

to obtain additional licenses.

Note: Each SIP call between two SIP endpoints requires two SIP trunks for the duration

of the call. The license file installed on the system controls the maximum permitted.

display syste

-parameters customer

ptions Page 2 of 11

OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 12000 0

Maximum Concurrently Registered IP Stations: 18000 11

Maximum Administered Remote Office Trunks: 12000 0

Maximum Concurrently Registered Remote Office Stations: 18000 0

Maximum Concurrently Registered IP eCons: 414 0

Max Concur Registered Unauthenticated H.323 Stations: 100 0

Maximum Video Capable Stations: 18000 0

Maximum Video Capable IP Softphones: 18000 0

Maximum Administered SIP Trunks: 24000 172

Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0

Maximum Number of DS1 Boards with Echo Cancellation: 522 0

Maximum TN2501 VAL Boards: 128 0

Maximum Media Gateway VAL Sources: 250 1

Maximum TN2602 Boards with 80 VoIP Channels: 128 0

Maximum TN2602 Boards with 320 VoIP Channels: 128 0

Maximum Number of Expanded Meet-me Conference Ports: 300 0

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

8 of 44

11xx12xx-SM

5.2. IP Codec Set

This section describes the steps for administering an IP codec set, which is used in the IP

network region when Communication Manager communicates with the SIP telephones via

Session Manager.

Step Description

Enter the change ip-codec-set n command, where n is a number between 1 and 7,

inclusive. IP codec sets are used in Section 5.3 for configuring an IP network region to

specify which codec sets may be used within and between network regions. For the

compliance testing, G.722-64K, G.711MU, G.729A, and G.729AB were tested. If only

one codec should be used, then only specify the one that is to be used.

Note: for G.729 interoperability between Avaya 1100- and 1200-Series IP Deskphones

and Avaya 9600 Series SIP Telephones, the configuration file settings for all telephones

should match that in the IP codec set. For the sample configuration shown below, the

9600 SIP Telephone configuration file should include the line: SET ENABLE_G729 “2”,

and the 1100- and 1200-Series IP Deskphone configuration files should include the line

“G729_ENABLE_ANNEXB YES”. See References [7, 8] and Section 7.2.

change ip-codec-set 1 Page 1 of 2

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.722-64K 2 20

2: G.711MU n 2 20

3: G.729AB n 2 20

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

9 of 44

11xx12xx-SM

5.3. IP Network Region

This section describes the steps for administering an IP network region, which is used when

Communication Manager communicates with the SIP telephones via Session Manager.

Step Description

Enter the change ip-network-region n command, where n is a number between 1 and

250 inclusive and configure the following as shown in the display screen below:

 Authoritative Domain – Set to avaya.com for the sample configuration. This

should match the SIP Domain value configured in Session Manager.

 Intra-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio

connectivity between endpoints registered to Communication Manager or Session

Manager in the same IP network region.

 Inter-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio

connectivity between endpoints registered to Communication Manager or Session

Manager in different IP network regions.

 Codec Set – Set the codec set number as provisioned in Section 5.2.

change ip-network-region 1 Page 1 of 20

IP NETWORK REGION

Region: 1

Location: Authoritative Domain: avaya.com

Name: CM and SIP Phones

MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes

Codec Set: 1 Inter-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? y

UDP Port Max: 65535

DIFFSERV/TOS PARAMETERS

Call Control PHB Value: 0

Audio PHB Value: 46

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 6

Audio 802.1p Priority: 6

Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS

5.4. IP Node Names

This section describes the steps for administering a node name for Session Manager to be used in

the configuration of the SIP signaling group.

Step Description

Use the change node-names ip command to add a new node name for Session Manager.

change node-names ip Page 1 of 2

IP NODE NAMES

Name IP Address

SM1 10.1.2.70

default 0.0.0.0

procr 10.1.2.90

procr6 ::

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

10 of 44

11xx12xx-SM

5.5. SIP Signaling Group

This section describes the steps for administering a signaling group for communication between

Communication Manager and Session Manager.

Step Description

Enter the command add signaling-group n, where n is an available signaling group and

configure the following as shown in the display screen below:

 Group Type – Set to sip.

 Transport Method – Set to tls.

 IMS Enabled – Set to n.

 Near-end Node Name - Set to procr.

 Near-end Listen Port - Defaults to 5061 for TLS.

 Far-end Node Name - Set to the node name configured in Section 5.4.

 Far-end Listen Port - Defaults to 5061 for TLS.

 Far-end Network Region - Set to the Region configured in Section 5.3.

 Far-end Domain - Set to avaya.com for the sample configuration. This should

match the SIP Domain value configured in Session Manager.

 Direct IP-IP Audio Connections – Set to y.

signaling-group 60 Page 1 of 1

SIGNALING GROUP

Group Number: 60 Group Type: sip

IMS Enabled? n Transport Method: tls

Q-SIP? n SIP Enabled LSP? n

IP Video? n Enforce SIPS URI for SRTP? y

Peer Detection Enabled? y Peer Server:

Near-end Node Name: procr Far-end Node Name: SM1

Near-end Listen Port: 5061 Far-end Listen Port: 5061

Far-end Network Region: 1

Far-end Domain: avaya.com

Bypass If IP Threshold Exceeded? n

Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n

DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y

Session Establishment Timer(min): 3 IP Audio Hairpinning? n

Enable Layer 3 Test? n Initial IP-IP Direct Media? n

H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 10

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

11 of 44

11xx12xx-SM

5.6. SIP Trunk Group

This section describes the steps for administering a trunk group for communication between

Communication Manager and Session Manager.

Step Description

Issue the command add trunk-group n, where n is an unallocated trunk group and

configure the following as shown in the display screen below:

 Group Type – Set to the Group Type field to sip.

 Group Name – Enter any descriptive name.

 TAC (Trunk Access Code) – Set to any available trunk access code.

 Service Type – Set to tie.

 Signaling Group – Set to the Group Number field value configured in Section

5.5. (i.e., 60)

 Number of Members – Allowed values are between 0 and 255. Set to a value

large enough to accommodate the number of SIP telephone extensions being used.

Note: Each SIP call between two SIP endpoints (whether internal or external) requires

two SIP trunk members for the duration of the call. The license file installed on the system

controls the maximum permitted.

add trunk-group 60 Page 1 of 21

TRUNK GROUP

Group Number: 60 Group Type: sip CDR Reports: y

Group Name: SM1 COR: 1 TN: 1 TAC: 160

Direction: two-way Outgoing Display? n

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Member Assignment Method: auto

Signaling Group: 60

Number of Members: 120

Proceed to Page 3 and set Numbering Format to private.

change trunk-group 60 Page 3 of 21

TRUNK FEATURES

ACA Assignment? n Measured: none

Maintenance Tests? y

Numbering Format: private

UUI Treatment: service-provider

Replace Restricted Numbers? n

Replace Unavailable Numbers? N

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

12 of 44

11xx12xx-SM

5.7. Define System Features

This section describes the steps for administering system wide call features and options related to

OPS in Communication Manager.

Step Description

Use the change system-parameters features command and navigate to Page 18 to

administer system wide features for the SIP telephones. Those related to features listed in

Table 2 are shown outlined in red. These are all standard Communication Manager

features.

change syste

-parameters features Page 18 of 19

FEATURE-RELATED SYSTEM PARAMETERS

INTERCEPT TREATMENT PARAMETERS

Invalid Number Dialed Intercept Treatment: tone

Invalid Number Dialed Display:

Restricted Number Dialed Intercept Treatment: tone

Restricted Number Dialed Display:

Intercept Treatment On Failed Trunk Transfers? n

WHISPER PAGE

Whisper Page Tone Given To: paged

change system-parameters features Page 19 of 19

FEATURE-RELATED SYSTEM PARAMETERS

IP PARAMETERS

Direct IP-IP Audio Connections? y

IP Audio Hairpinning? y

Synchronization over IP? n

CALL PICKUP

Maximum Number of Digits for Directed Group Call Pickup: 4

Call Pickup on Intercom Calls? y Call Pickup Alerting? n

Temporary Bridged Appearance on Call Pickup? y Directed Call Pickup? y

Extended Group Call Pickup: simple

Enhanced Call Pickup Alerting? N

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

13 of 44

11xx12xx-SM

5.8. Define the Dial Plan

This section describes the steps for administering the dial plan, including overall dial plan

format, Feature Access Codes (FACs), and Feature Name Extensions (FNEs).

Step Description

Use the change dialplan analysis command to define the dial plan formats used in the

system. This includes all telephone extensions, Feature Name Extensions (FNEs), and

Feature Access Codes (FACs). To define the FNEs for the features listed in Table 2, a

Feature Access Code (FAC) must also be specified for the corresponding feature. In the

sample configuration, telephone extensions are five digits long and begin with 3, FNEs

are five digits beginning with 7, and the FACs have formats as indicated with Call Type

“fac”. Note: a FAC of “8” was used for AAR routing by a voice mail hunt group, the

configuration for which is not included in these Application Notes. See Reference [11]

for more information.

change dialplan analysis Page 1 of 12

DIAL PLAN ANALYSIS TABLE

Location: all Percent Full: 2

Dialed Total Call Dialed Total Call Dialed Total Call

String Length Type String Length Type String Length Type

0 3 fac

1 3 dac

2 5 ext

3 5 ext

4 5 ext

5 5 ext

6 3 fac

7 5 ext

8 1 fac

9 1 fac

* 2 fac

# 2 fac

Use the change private-numbering command to add an entry as shown below for the

calling extensions that will be using the trunk to Session Manager. The entry specifies the

format that the calling number will have in outgoing calls. Set Ext Len to the length of

the calling extensions, Ext Code to an initial set of digits that covers the extension range,

Trk Grp(s) to the trunk group number defined in Section 5.6, and Total Len to the

length of the calling extensions. In the sample configuration, the extension is sent

unchanged. Note: if the Trk Grp(s) field is left blank, as in the first entry, this formatting

will be applied to all trunk groups in the system.

change private-numbering 0 Page 1 of 2

NUMBERING - PRIVATE FORMAT

Ext Ext Trk Private Total

Len Code Grp(s) Prefix Len

5 2 5 Total Administered: 3

5 3 60 5 Maximum Entries: 540

5 4 60 5

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

14 of 44

11xx12xx-SM

Use change feature-access-codes to define the access codes for the FNEs highlighted in

red. The following screens have been abbreviated to highlight those FACs involved in

supporting the FNEs and the AAR FAC.

change feature-access-codes Page 1 of 10

FEATURE ACCESS CODE (FAC)

Answer Back Access Code: 625

Attendant Access Code:

Auto Alternate Routing (AAR) Access Code: 8

Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2:

Automatic Callback Activation: *5 Deactivation: #5

Call Forwarding Activation Busy/DA: *2 All: 612 Deactivation: #2

Call Forwarding Enhanced Status: Act: Deactivation:

Call Park Access Code: 624

Call Pickup Access Code: *6

CAS Remote Hold/Answer Hold-Unhold Access Code: #6

change feature-access-codes Page 2 of 10

FEATURE ACCESS CODE (FAC)

Contact Closure Pulse Code:

Directed Call Pickup Access Code: 654

Directed Group Call Pickup Access Code:

Emergency Access to Attendant Access Code:

EC500 Self-Administration Access Codes:

Enhanced EC500 Activation: 660 Deactivation: 661

Enterprise Mobility User Activation: Deactivation:

Extended Call Fwd Activate Busy D/A All: Deactivation:

Extended Group Call Pickup Access Code: 641

change feature-access-codes Page 3 of 10

FEATURE ACCESS CODE (FAC)

PASTE (Display PBX data on Phone) Access Code:

Personal Station Access (PSA) Associate Code: Dissociate Code:

Per Call CPN Blocking Code Access Code: 615

Per Call CPN Unblocking Code Access Code: 616

Posted Messages Activation: Deactivation:

Priority Calling Access Code: *7

Program Access Code: *0

Refresh Terminal Parameters Access Code: 694

Remote Send All Calls Activation: Deactivation:

Self Station Display Activation:

Send All Calls Activation: *3 Deactivation: #3

Station Firmware Download Access Code:

change feature-access-codes Page 4 of 10

FEATURE ACCESS CODE (FAC)

Transfer to Voice Mail Access Code: #9

Trunk Answer Any Station Access Code:

User Control Restrict Activation: 691 Deactivation: 692

Voice Coverage Message Retrieval Access Code:

Voice Principal Message Retrieval Access Code:

Whisper Page Activation Access Code: 620

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

15 of 44

11xx12xx-SM

FNEs are defined using the change off-pbx-telephone feature-name-extensions set n

command, where n is a number between 1 and 99 and will default to 1 if n is not

specified. This command is used to support both SIP telephones and Extension to

Cellular. The highlighted fields correspond to those features listed as supported in Table

2. The fields that have been left blank correspond to those more appropriate for Extension

to Cellular.

change off-pbx-telephone feature-name-extensions set 1 Page 1 of 2

EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME

Set Name:

Active Appearance Select: 70030

Automatic Call Back: 70003

Automatic Call-Back Cancel: 70004

Call Forward All: 70005

Call Forward Busy/No Answer: 70006

Call Forward Cancel: 70007

Call Park: 70008

Call Park Answer Back: 70009

Call Pick-Up: 70010

Calling Number Block: 70012

Calling Number Unblock: 70013

Conditional Call Extend Enable:

Conditional Call Extend Disable:

Conference Complete:

Conference on Answer: 70011

Directed Call Pick-Up: 70014

Drop Last Added Party: 70015

change off-pbx-telephone feature-name-extensions set 1 Page 2 of 2

EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME

Exclusion (Toggle On/Off): 70016

Extended Group Call Pickup: 70025

Held Appearance Select: 70017

Idle Appearance Select:

Last Number Dialed: 70019

Malicious Call Trace: 70029

Malicious Call Trace Cancel: 70021

Off-Pbx Call Enable: 70027

Off-Pbx Call Disable: 70028

Priority Call: 70000

Recall:

Send All Calls: 70001

Send All Calls Cancel: 70002

Transfer Complete:

Transfer On Hang-Up: 70022

Transfer to Voice Mail: 70023

Whisper Page Activation: 70026

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

16 of 44

11xx12xx-SM

5.9. Specify Class of Service (COS) and Class Of Restriction (COR)

This section describes the steps for administering the COS and COR, which affects what calling

features and feature options are permitted for defined groups of telephone users.

Step Description

Use the change cos-group n command, where n is a class of service group number, to set

the appropriate service permissions to support the corresponding features (shown in

outlined in red). For the sample configuration, COS group 1 was used. On Page 2, set

the value of VIP Caller to “y” only if all calls made by telephones with this COS should

be priority calls. Note that priority can be requested on a call-by-call basis by using the

Priority Call FNE (see Section 5.8). Priority call indication (e.g., distinctive ring and

display of “Priority”) is only supported on Avaya Digital and 9600 Series IP telephones

(H.323 and SIP).

change cos-group 1 Page 1 of 2

CLASS OF SERVICE COS Group: 1 COS Name:

0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

Auto Callback n y y n y n y n y n y n y n y n

Call Fwd-All Calls n y y y y n n y y n n y y n n n

Data Privacy n n n y n y y y y n n n n y y y

Priority Calling n y n n n n n n n y y y y y y n

Console Permissions y y y n n n n n n n n n n n n n

Off-hook Alert n n n n n n n n n n n n n n n n

Client Room n n n n n n n n n n n n n n n n

Restrict Call Fwd-Off Net n n y y y y y y y y y y y y y y

Call Forwarding Busy/DA n y n n n n n n n n n n n n n n

Personal Station Access (PSA) n n n n n n n n n n n n n n n n

Extended Forwarding All n y n n n n n n n n n n n n n n

Extended Forwarding B/DA n y n n n n n n n n n n n n n n

Trk-to-Trk Transfer Override n y n n n n n n n n n n n n n n

QSIG Call Offer Originations n n n n n n n n n n n n n n n n

Contact Closure Activation n n n n n n n n n n n n n n n n

change cos-group 1 Page 2 of 2

CLASS OF SERVICE

0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

VIP Caller n n n n n n n n n n n n n n n n

Masking CPN/Name Override n n n n n n n n n n n n n n n n

Call Forwarding Enhanced y y y y y y y y y y y y y y y y

Priority Ip Video n n n n n n n n n n n n n n n n

Ad-hoc Video Conferencing n n n n n n n n n n n n n n n n

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

17 of 44

11xx12xx-SM

Use the change cor n command, where n is a class of restriction number, to enable

applicable calling features. To use the Directed Call Pickup feature, the Can Use

Directed Call Pickup and Can Be Picked Up By Directed Call Pickup fields must be

set to “y” for the affected stations. In the sample configuration, the telephones were

assigned to COR 1. Note that Page 4 can be used to implement a form of centralized call

screening for groups of stations and trunks.

change cor 1 Page 1 of 23

CLASS OF RESTRICTION

COR Number: 1

COR Description: Trunk

FRL: 0 APLT? y

Can Be Service Observed? y Calling Party Restriction: none

Can Be A Service Observer? y Called Party Restriction: none

Time of Day Chart: 1 Forced Entry of Account Codes? n

Priority Queuing? n Direct Agent Calling? y

Restriction Override: none Facility Access Trunk Test? n

Restricted Call List? n Can Change Coverage? n

Access to MCT? y Fully Restricted Service? n

Group II Category For MFC: 7 Hear VDN of Origin Annc.? n

Send ANI for MFE? n Add/Remove Agent Skills? n

MF ANI Prefix: Automatic Charge Display? n

Hear System Music on Hold? y PASTE (Display PBX Data on Phone)? n

Can Be Picked Up By Directed Call Pickup? y

Can Use Directed Call Pickup? y

Group Controlled Restriction: inactive

change cor 1 Page 4 of 23

CLASS OF RESTRICTION

CALLING PERMISSION (Enter "y" to grant permission to call specified COR)

0? y 15? y 30? y 44? y 58? y 72? y 86? y

1? y 16? y 31? y 45? y 59? y 73? y 87? y

2? y 17? y 32? y 46? y 60? y 74? y 88? y

3? y 18? y 33? y 47? y 61? y 75? y 89? y

4? y 19? y 34? y 48? y 62? y 76? y 90? y

5? y 20? y 35? y 49? y 63? y 77? y 91? y

6? y 21? y 36? y 50? y 64? y 78? y 92? y

7? y 22? y 37? y 51? y 65? y 79? y 93? y

8? y 23? y 38? y 52? y 66? y 80? y 94? y

9? y 24? y 39? y 53? y 67? y 81? y 95? y

10? n 25? y 40? y 54? y 68? y 82? y 96? y

11? y 26? y 41? y 55? y 69? y 83? y 97? y

12? y 27? y 42? y 56? y 70? y 84? y 98? y

13? y 28? y 43? y 57? y 71? y 85? y 99? y

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

18 of 44

11xx12xx-SM

5.10. SIP Stations

This section describes the steps for administering OPS stations in Communication Manager and

associating the OPS station extensions with the telephone numbers of the Avaya 1100-and 1200-

Series IP Deskphones. The configuration is the same for all phones except for the desired

number of call appearances as detailed in Step 3. Note that the corresponding users must be

configured in Session Manager. There are two methods to sequence these steps:

1. Configure the station and off-PBX-station forms for each user in Communication

Manager. Then configure the corresponding user in Session Manager, being sure to

check the “Use Existing Stations” box (see Section 6).

2. Configure the user in Session Manager, being sure to leave the “Use Existing Stations”

box unchecked (see Section 6). Session Manager will automatically generate the

corresponding station and off-PBX-station information in Communication Manager.

Then use the change station command in Communication Manager to add other

configuration data, such as Coverage Path, MWI Served User Type, and additional call

appearances, if needed.

Method 2 was used in the sample configuration. For method 1, perform the following steps for

each user; then follow the steps in Section 6. For method 2, follow the steps in Section 6 first;

then use change station n to modify any station parameters as described below using the station

form in this section as a guide.

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

19 of 44

11xx12xx-SM

Step Description

Enter the add station n command, where n is an available extension in the dial plan, to

administer an OPS station. On Page 1 of the form configure the following fields as shown

in the display screen below:

 Type – Set to 9630SIP.

 Port – Leave blank. (Once the form is submitted, a virtual port is assigned, e.g.,

S00022)

 Name – Enter any descriptive name.

 Coverage Path – Enter the coverage path number defined for this telephone (e.g.,

for coverage to voice mail).

add station 30043 Page 1 of 6

STATION

Extension: 30043 Lock Messages? n BCC: 0

Type: 9630SIP Security Code: TN: 1

Port: Coverage Path 1: 60 COR: 1

Name: 1165E Coverage Path 2: COS: 1

Hunt-to Station:

STATION OPTIONS

Time of Day Lock Table:

Loss Group: 19

Message Lamp Ext: 30043

Display Language: english Button Modules: 0

Survivable COR: internal

Survivable Trunk Dest? y IP SoftPhone? n

IP Video? N

FS; Reviewed:

SPOC 09/07/2010

Solution & Interoperability Test Lab Application Notes

20 of 44

11xx12xx-SM

Proceed to Page 2 of the form. Set MWI Served User Type to sip-adjunct.

add station 30043 Page 2 of 6

STATION

FEATURE OPTIONS

LWC Reception: spe

LWC Activation? y Coverage Msg Retrieval? y

Auto Answer: none

CDR Privacy? n Data Restriction? n

Idle Appearance Preference? n

Per Button Ring Control? n Bridged Idle Line Preference? n

Bridged Call Alerting? n

Active Station Ringing: single

H.320 Conversion? n Per Station CPN - Send Calling Number?

EC500 State: enabled

MWI Served User Type: sip-adjunct

Coverage After Forwarding? s

Direct IP-IP Audio Connections? y

Emergency Location Ext: 30043 Always Use? n IP Audio Hairpinning? N

Proceed to Page 4 of the form and add the desired number of call-appr entries in the

BUTTON ASSIGNMENTS section. This governs how many concurrent calls can be

supported. Avaya 1100-Series IP Deskphones have the capability of handling 11 call

appearances (10 for the 1200-Series), but display only one local call appearance button

when idle (see display in Section 7.4 Step 3). So the number of entries shown below are

not required to match that displayed on the telephone. Three are configured here to

support conferencing scenarios.

add station 30043 Page 4 of 6

STATION

SITE DATA

Room: Headset? n

Jack: Speaker? n

Cable: Mounting: d

Floor: Cord Length: 0

Building: Set Color:

ABBREVIATED DIALING

List1: List2: List3:

BUTTON ASSIGNMENTS

1: call-appr 5:

2: call-appr 6:

3: call-appr 7:

4: 8:

Page is loading ...

Avaya 1200-Series Application Note

Brand: Avaya

Model: 1200-Series

Category: Networking

Type: Application Note

Download PDF

report

Avaya 1200-Series Application Note

Avaya 1200-Series Application Note

Related papers

Other documents